JP3186315B2 - Signal compression device, signal decompression device, signal transmission device, signal reception device, and signal transmission / reception device - Google Patents

Abstract

Audio signals are compressed by selecting either a first compression mode providing high-quality sound and a relatively long processing time or a second compression mode having a shorter processing time in accordance with the contents of the audio input signals, for example, whether the signals are audio signals such as music or voice signals such as an announcement. By providing selectable compression time periods music can be enjoyed in high-quality sound, while the announcement of a message can be delivered satisfactorily without the loss of any information.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a signal compression device, a signal expansion device, a signal transmission device, a signal reception device, and a signal transmission / reception device for compressing and transmitting a signal, and more particularly, to high-quality audio signals and voice signals. The present invention relates to a signal compression device, a signal decompression device, a signal transmission device, a signal reception device, and a signal transmission / reception device that perform compression and transmission / reception while maintaining or shortening the time delay associated with processing.

[0002]

2. Description of the Related Art There are various known high-efficiency coding methods for audio signals. For example, non-blocking methods in which an audio signal or the like on a time axis is divided into a plurality of frequency bands and coded without being blocked. Frequency band division system,
Band division coding (sub-band coding: SB
C) or a blocking frequency band division method of converting a signal on the time axis into a signal on the frequency axis (spectral conversion), dividing the signal into a plurality of frequency bands, and encoding each band. Can be mentioned.

Further, a high-efficiency coding method combining the above-described band division coding and transform coding has been considered. In this case, for example, after performing band division by the above-mentioned band division coding, The spectrum of the signal of each band is converted into a signal on the frequency axis, and coding is performed for each of the spectrum-converted bands. As a filter for the band division, there is a so-called QMF filter, for example.

Here, as the above-mentioned spectral transform, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a fast Fourier transform (FFT), a cosine transform (DCT), a modified DCT transform ( There is a spectrum transformation that transforms a time axis into a frequency axis by performing (MDCT) or the like. MDC
About T ICASSP 1987, Subband / Transform Codin
g Using Filter Bank Designs Based on Time Domain A
liasing Cancellation, JPPrincen, ABBradley, Un
iv. of Surrey Royal Melbourne Inst. of Tech.

[0005] By quantizing the signal divided for each band by the filter or the spectrum conversion as described above,
It is possible to control the band in which the quantization noise is generated, and to perform aurally more efficient coding by utilizing properties such as a masking effect. Further, if the normalization is performed for each band, for example, with the maximum value of the absolute value of the signal component in the band before performing the quantization, more efficient coding can be performed.

[0006] As a frequency division width for quantizing each frequency component divided into frequency bands, for example, band division is performed in consideration of human auditory characteristics. That is, an audio signal may be divided into a plurality of bands (for example, 25 bands) with a bandwidth generally called a critical band (critical band) such that the bandwidth becomes wider as the band becomes higher. Also,
At this time, when encoding data for each band, predetermined bits are allocated to each band, or encoding is performed by adaptive bit allocation (bit allocation) for each band. For example, when the coefficient data obtained by the MDCT processing is encoded by the bit allocation, adaptively to the MDCT coefficient data of each band obtained by the MDCT processing of each block, Encoding is performed with the allocated number of bits.

[0007] Such high-efficiency compression coding is suitable for applications in which high-quality audio signals are transmitted without expanding the transmission band. For example, when configuring a system in which audio signals can be individually listened to through a plurality of seats in a vehicle such as an aircraft with headphones or earphones, and audio signals of a large number of channels can be freely selected in each seat, In order to realize multi-channel transmission without deteriorating signal quality, it is desirable to use the above-described high-efficiency compression coding.

[0008]

By the way, in the high-efficiency compression coding system which has good sound quality while maintaining a high compression rate as described above, the compression processing (encoding processing) and the decompression processing (decoding processing) are relatively difficult. In many cases, a long time is required, and a time lag occurs between a signal transmitted straight and not subjected to the compression / expansion processing. Therefore, for example, when the present invention is applied to a headphone listening type audio system provided in a seat or the like in a vehicle such as an aircraft, a sound from a so-called PA speaker provided on a wall surface or a ceiling (a captain's announcement or the like). ) Occurs, making it difficult to hear.

That is, for example, when a message from the captain is to be announced in an aircraft, sound is output from a so-called PA speaker, and the channel selection state of the audio signal for each seat is forcibly changed to a predetermined PA signal. The control is switched to the channel so that the message can be reliably transmitted to the passengers who are listening through headphones or the like. However, while the sound from the PA speaker is normally transmitted in analog form and reproduced straight (without time delay), if the sound from the headphones is delayed by the compression / expansion processing time, these Because of the time lag between the sounds of
There is a problem that a so-called echo occurs and the message becomes difficult to hear, or a part of a word is missing when the channel is forcibly switched to the PA channel.

[0010] Further, in the case of sending a compressed voice to the in-flight speaker, since the person who speaks the message listens to his / her own voice broadcast from the in-flight speaker, a problem such as confusion occurs if the time delay is large. There is a fear.

The present invention has been made in view of such circumstances, and while audio signals such as music have high sound quality, audio signals such as announcement messages have a short compression / expansion processing time and a small time delay. It is an object of the present invention to provide a signal compression device, a signal expansion device, a signal transmission device, a signal reception device, and a signal transmission / reception device.

[0012]

A signal compression apparatus according to the present invention has a first compression mode in which a predetermined time is required for signal compression processing, and a second compression mode in which the signal compression processing time is shorter than the first compression mode. A signal compression device having a signal compression section that can be switched between compression modes, wherein an audio signal such as music and an audio signal such as an announcement are used as signal contents to be compressed, and the audio signal is the first signal. The above-mentioned problem is solved by switching and selecting the first compression mode and the second compression mode so that the audio signal is compressed in the compression mode and the audio signal is compressed in the second compression mode.

In order to solve the above-mentioned problems, a signal decompression device according to the present invention has a first decompression mode that requires a predetermined time for signal decompression processing of a compressed signal, and a signal decompression processing time of the first decompression time. A signal decompression device having a signal decompression unit switchable between a second decompression mode shorter than the mode,
An audio signal such as music and an audio signal such as an announcement are used as the signal contents to be expanded. The audio signal is expanded in the first expansion mode, and the audio signal is expanded in the second expansion mode. The first and second expansion modes are switched and selected.

According to another aspect of the present invention, there is provided a signal transmitting apparatus including a signal compressing apparatus for compressing a signal by the signal compressing apparatus and transmitting the compressed signal via a transmission path. The compression device has a signal compression unit that can be switched between a first compression mode requiring a predetermined time for signal compression processing and a second compression mode in which the signal compression processing time is shorter than the first compression mode. An audio signal such as music and an audio signal such as an announcement are used as signal contents to be compressed, and the audio signal is compressed in the first compression mode and the audio signal is compressed in the second compression mode. Further, the present invention is characterized in that the first compression mode and the second compression mode are switched and selected.

In order to solve the above-mentioned problems, a signal receiving apparatus according to the present invention includes a signal receiving apparatus which receives a compressed signal via a transmission line and expands the compressed signal. The decompression device includes a first decompression mode that requires a predetermined time for signal decompression processing of a compressed signal, and a second decompression processing time that is shorter than the first decompression mode.
A signal decompression device having a signal decompression unit that can be switched between the decompression mode and the decompression mode, wherein the signal content to be decompressed uses an audio signal such as music and a voice signal such as an announcement, and the audio signal is the first signal. Stretch in stretch mode,
The audio signal is switched and selected between the first expansion mode and the second expansion mode so that the audio signal is expanded in the second expansion mode.

In order to solve the above-mentioned problems, a signal transmitting / receiving apparatus according to the present invention includes a signal compressing apparatus, and a signal transmitting apparatus for compressing a signal with the signal compressing apparatus and transmitting the compressed signal via a transmission path; And a signal decompression device that receives a compressed signal via the first signal transmission device and decompresses the compressed signal, the signal compression device comprising: a first compression mode that requires a predetermined time for signal compression processing; A signal compression unit capable of switching between a second compression mode and a processing time shorter than the first compression mode, wherein the signal decompression device requires a predetermined time for signal decompression processing of the compressed signal; A signal decompression unit that can be switched between a decompression mode and a second decompression mode in which the signal decompression processing time is shorter than the first decompression mode, wherein the signal content to be compressed or decompressed is audio such as music. Faith And an audio signal such as an announcement, the audio signal is compressed or expanded in the first compression or expansion mode, and the audio signal is compressed or expanded in the second compression or expansion mode. The compression or expansion mode and the second compression or expansion mode are switched and selected.

Here, the first compression or decompression mode is a mode for performing high-efficiency encoding in which bit allocation for each band divided in the frequency axis direction is adaptively controlled using human auditory characteristics. Preferably, the second compression or decompression mode is an adaptive difference PCM mode.

[0018]

[Function] For applications requiring high sound quality such as listening to music, processing is performed in the first compression or decompression mode with little deterioration in sound quality even if the processing time is slightly longer, and a time delay such as an announcement is reduced. When it is desired to perform the processing in the second compression or decompression mode having a short processing time, the signal can be transmitted and received in a mode most suitable for the signal to be transmitted and received.

[0019]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Preferred embodiments according to the present invention will be described below with reference to the drawings.

FIG. 1 is a block diagram of a signal transmission / reception system to which a signal compression device, a signal expansion device, a signal transmission device, a signal reception device, and a signal transmission / reception device according to an embodiment of the present invention are applied.

In FIG. 1, an audio or voice signal supplied to an input terminal 1 is converted into a digital signal by an A / D converter 2, and then a first compression mode requiring a predetermined time for signal compression processing. A compression circuit 3a for performing compression and encoding in (first compression method) and a compression circuit for performing compression and encoding in a second compression mode (second compression method) in which the signal compression processing time is shorter than the first compression mode 3b. Output signals from these compression circuits 3a and 3b are sent to selected terminals a and b of the changeover switch 4, respectively. An output signal from the changeover switch 4 is sent to a transmission circuit 5 including a modulator, a multiplexer and the like, converted into a signal for transmission, and sent out via an output terminal 6.

The signal transmitted from the output terminal 6 and supplied to the input terminal 7 is supplied to a receiving circuit 8 including a demodulator and a demultiplexer. The output signal from the receiving circuit 8 is
A decompression circuit 9a that performs decompression decoding in the first decompression mode corresponding to the first compression mode, and a decompression circuit 9b that performs decompression decoding in the second decompression mode corresponding to the second compression mode. They are sent and decompressed respectively. Output signals from these expansion circuits 9a and 9b are sent to selected terminals a and b of the changeover switch 10, respectively. The output signal from the changeover switch 10 is converted into an analog signal by the D / A converter 11 and then taken out via the output terminal 12.

Here, the compression circuits 3a and 3b and the expansion circuit 9
Although a and 9b may be configured by hardware, they are preferably realized by software using a so-called DSP or the like. In this case, switching between the compression circuits 3a and 3b and switching between the expansion circuits 9a and 9b are performed. Is realized by switching the processing algorithm.

In the configuration shown in FIG. 1, A /
The time required for the conversion processing in the D converter 2 is Ta, and the time required for the compression processing in the compression circuits 3a and 3b is T.
_b1, and T _b2, and so on to, Td the time required for the duration of the transmitting circuit 5 Tc, the receiving circuit 8, decompression circuit 9a,
Assuming that the required time at 9b is T _e1 and T _e2 , respectively, and the required time at the D / A converter 11 is Tf, before the signal input to the input terminal 1 is taken out from the output terminal 12, FIG. A delay of the indicated time _TDL will occur. This figure 2
“A” to “f” represent the respective processes in the A / D converters 2 to D / A converter 11, respectively.

Now, as a specific example of the first compression mode, a so-called ATRAC (Adaptive TRansform Acoustic Acoustic) used for signal compression of a so-called MD (mini-disc).
A case will be described in which an AD (adaptive difference) PCM method adopted for signal compression of so-called CD-I or the like is used as a specific example of the second compression mode using a stic coding (stic coding) method.

In this algorithm of the so-called ATRAC system, 512 samples are collectively compressed as one sound frame, and data of 512 samples is generated at a time on the decompression side. A delay of four sound frames occurs. Further, when a multiplexer or demultiplexer IC is used for transmission and reception, 1 mse
c processing time, and the above A / D, D
A delay also occurs in the digital filter inside the / A converter. Here, as the actually measured values when the sampling frequency fs is 32 kHz, the times Ta and Tf required for the A / D and D / A conversions are each about 32 μsec, and the times Tb and Te (the first In the compression mode, T _b1 and T _e1 ) are each about 32 msec, so the delay time T _DL is about 75 msec. The sound of a video such as a movie shown on an airplane such as an aircraft is heard with a delay of about 75 msec. Generally, a so-called lip sync (synchronization between video and sound) is one sound.
Since it is permissible up to about 20 msec, it is understood that there is no problem.

However, a so-called PA (Pa), for example, which announces a message from the captain in an aircraft.
At the time of overriding, for example, the speaker listens to his / her own voice broadcast from the in-flight speaker while speaking into the microphone, so this time delay is not allowed. It is also necessary to minimize the time delay before the sound is output to the speaker of the present invention. Therefore, the PA
At the time of the override, the delay time is shortened by switching to the second compression mode.

When the ADPCM method is used in the second compression mode, the time T required for the compression and decompression is used.
b and Te (T _b2 and T _{e2 in} the second compression mode) are each about 6 msec, and the delay time T _DL is about 15 msec.
msec. In the ADPCM in this example, it is necessary to add a range bit and a filter bit in addition to the audio data, and a checksum for detecting a transmission error is added to this to form a fine block (for example, 16 samples × 2 channels). Transmitted in units.

Accordingly, entertainment such as music can be enjoyed with high sound quality by the first compression mode, and at the time of PA, there is no loss of sound, which can contribute to a safe flight.

Next, a signal transmission system to which a signal compression device or a signal transmission device as one embodiment of the present invention as described above is applied will be described with reference to FIG. FIG. 3 is a block diagram showing a schematic configuration of a system in which, for example, audio signals are individually listened to at each seat in an aircraft using headphones or the like.

In FIG. 3, a transmission signal source 110 includes a video signal reproducing device 111 such as a so-called VTR (video tape recorder) or a video disk player.
An audio signal reproducing device 112 such as a so-called CD (compact disk) player or a tape recorder;
A computer data output device 113 such as V-game software is provided. In addition, a so-called PA (Passenger) having a function for in-flight broadcasting using a microphone, etc.
Address) A control device 116 is provided.

Of the output signals from these devices, audio or audio signals from the video signal reproducing device 111 and the audio signal reproducing device 112 and computer data from the computer data output device 113 are transmitted to the audio signal transmitting device 120. Have been. The video signal from the video signal reproducing device 111 is sent to another video signal transmitting device (not shown), and the audio signal from the PA control device 116 is directly sent to a speaker (not shown) provided on the wall or ceiling inside the device. (For example, as an analog audio signal) and the audio signal transmitting device 12
Also sent to 0.

The audio signal transmitting device 120 includes, for example, up to four audio signal transmitting units 121A and 121A.
1B, 121C, and 121D. One audio signal transmission unit 121 multiplexes, for example, an audio signal of 32 channels and a data signal of 8 channels (or an audio signal of 36 channels). The signal is modulated into an RF signal and output.
An RF mixing circuit that inputs and mixes an RF output signal from another audio signal transmission unit is provided in the RF modulation unit 122. Specifically, each of the audio signal transmission units 121A, 121B, 121C, 121D
RF signals of different carrier frequencies are frequency-multiplexed and output every time. In this case, the video signal reproducing device 111
The video signals of a plurality of channels may also be modulated at different carrier frequencies by an RF modulator (not shown), frequency-multiplexed with the RF modulation signal of the audio signal, and transmitted through a single coaxial cable.

The frequency-multiplexed RF signal is sent to an RF splitter 131 of a zone management unit 130 provided for each of the zones that divide the inside of the aircraft into zones.
1 to the zone management unit of the next zone via the RF amplifier 132. The RF signal input to the RF splitter 131 is split and transmitted for each column, which is a division area for dividing the zone. A plurality of seat units 140 are connected in series in one column.
The F signal is sent to the next seat unit 140 via the RF splitter 141 and the RF amplifier 142.

In one seat unit 140, for example, 3
Circuits for the seats are provided collectively, and the RF signal split by the RF splitter 141 is split into three RF signals by the RF splitter 144, and the three RF receivers 145
A, 145B, and 145C, respectively. The RF receiving section 145 extracts an RF signal in one band of the frequency-multiplexed signal, performs RF demodulation, extracts a signal of a desired channel from the demodulated signal, and outputs the signal. The signals output from each of the RF receivers 145A, 145B, and 145C are transmitted to a passenger-side control unit 146A,
46B and 146C, respectively. The passengers of each seat can connect to the control units 146A, 146B, and 146C with the headphones 147A, 147B, and 147C, respectively, and arbitrarily select and listen to an audio signal of a desired channel.

Next, the audio signal transmitting unit 121 on the audio signal transmitting side will be described with reference to FIG.

The audio signal transmitting unit 121 shown in FIG. 4 receives, for example, an audio signal of 32 channels and a data signal of 8 channels and has a frequency band of 6 channels.
It is converted into a signal of MHz and output. The audio signal transmission unit 121 is also called an audio multiplexer unit or AMUX because the audio signal is multiplexed and output.

[0038] The audio signal transmitting unit 121, and 32-channel audio signal AU ₁ ~AU ₃₂ is for example supplied in the form of a so-called balanced inputs,
Each signal is converted by the buffer amplifier 21 from a balanced input form to an unbalanced form signal. Buffer amplifier 2
The audio signals of 32 channels from 1 are grouped for every two channels (for example, stereo left and right channels), sent to an A / D converter 22 and converted into digital signals, sent to a signal compression circuit 23 and compressed respectively. It is encoded. In the signal compression circuit 23, for example, a high-efficiency compression encoding process that adaptively controls the bit allocation for each band divided in the frequency axis direction using a human auditory characteristic according to an input signal, or a so-called so-called AD
(Adaptive difference) PCM encoding processing and the like are performed.

The computer data DT _{1 to} DT ₈ such as 8-channel TV game software are input in a balanced manner according to, for example, the format of a so-called EIA-422 interface, and converted into unbalanced signals by the buffer amplifier 21, respectively. 24.

Here, as the signal compression circuit 23, a first compression mode for performing a high-efficiency encoding process which takes a long processing time using the above-mentioned human auditory characteristics or the like but has good reproduction sound quality, and an ADPCM or the like. A circuit capable of switching and selecting the second compression mode having a relatively short processing time is used,
When listening to audio signals such as music for each seat, switch to the first compression mode. When forcibly switching to the PA channel at the time of announcement of a message or the like from the captain, switch to the second compression mode. It is preferable to switch. This is because the first compression mode is preferable because high sound quality is desired when listening to music or the like. During an announcement or the like, the sound from the speakers provided on the wall or ceiling of the aircraft and the sound from the headphones are used. This is because the second compression mode in which the time difference between them is small is more preferable. The selection of the switching of the compression mode is performed by a control signal from the controller 30. As a specific example of the first compression mode, a so-called ATRAC (Adaptive T) adopted in signal compression of a so-called MD (mini disc) is used.
Ransform Acoustic Coding) method or so-called dcc
So-called PASC (Precision Adaptive Sub-b) used for signal compression of digital compact cassettes
and Coding) method.

The output signals for the above-mentioned 32 channels from the signal compression circuit 23 are grouped into 12 channels (6 pairs of 2 channels) in order from the signal corresponding to the AU ₁ of the original input audio signal, and the multiplexer. 25
A and 25B respectively, and the remaining AU _{25 to} AU
Eight channels corresponding to ₃₂ sent to the output signal and the output signal are grouped together by the multiplexer 25C from the digital data interface circuit 24 corresponding to the data (DT ₁ to DT ₈₎ of the eight channels (four pairs min) Can be The digital data interface circuit 2
4, the clock signal is taken out to the outside via a buffer amplifier.

Multiplexers 25A, 25B, 25C
Respectively handle signals conforming to the audio signal format of, for example, satellite broadcasting or television broadcasting signals of satellite communication. One transmission frame of a format conforming to this audio signal format is composed of 32 columns × 64 columns, and in one transmission frame, 128 bits of data per channel is composed of 12 channels (for example, audio 12 channels or audio 12 channels). 8 channels +
8 channels of computer data) and what is called BC
At least an H error correction code (for example, a (63, 50) BCH code) and a preamble such as a frame synchronization and a control code are included. Data is output from one multiplexer 25 at a rate of, for example, 2.048 Mbps, and output data from each of the multiplexers 25A, 25B, 25C is sent to the time division multiplexing circuit 26 and time division multiplexed, thereby obtaining 6.144 Mbps. Is output as data of the rate. The digital signal output from the time division multiplexing circuit 26 is waveform-equalized by the equalizer 27,
The signal is compressed to a frequency band of Hz and sent to the RF modulator 122.

In the RF modulator 122, the output signal from the equalizer 27 is modulated by the intermediate frequency modulator 31 to become a predetermined intermediate frequency (IF) signal, amplified by the IF amplifier 32, and multiplied by the multiplier (modulator). ) 33. A predetermined RF from a PLL (phase locked loop) circuit 34 is input to the multiplier 33.
A carrier signal of a frequency is supplied, and the frequency of the output signal of the PLL circuit 34 is controlled by a controller 30 using a CPU or the like. Multiplier 33
Is output from a BPF (Band Pass Filter) 35
Through an RF amplifier 36 to an RF combiner (synthesizer or mixer) 37. This RF combiner 37 is connected to an external RF input terminal 126 via
For example, the RF from the other audio signal transmitting unit
A signal is being supplied. By making the RF carrier frequency of the RF modulator 122 different from the carrier frequency of the external RF input signal, the RF combiner 37 can perform frequency multiplexing. RF combiner 37
Is output to an output terminal 1 via an attenuator 38.
25.

That is, the RF modulation section 122
The frequency is determined based on, for example, three program discrete signals supplied to the controller 30, by controlling the frequency division ratio or the prescaler value in the PLL circuit 34 by the controller 30. The frequency is variably controlled, and the carrier frequency of the RF modulation signal from the multiplier 33 is variably controlled. That is, RF is generated by the program discrete signal.
The carrier frequency of the signal is determined so that it does not match (is different from) the carrier frequency of the RF signal from the other audio signal transmitting unit. Also, external RF
When there is no RF signal input from the input terminal 126, the controller 30 is controlled by the external RF input terminal 126 according to a termination control signal supplied to the controller 30.
Are grounded, and the RF output terminal 125 is terminated.

Next, the audio signal transmitting unit 1
The self-diagnosis function 21 will be described. The output signal from the time-division multiplexing circuit 26 is demultiplexed by the demultiplexer 28 and sent to the controller 30 using a CPU or the like, thereby determining whether or not the multiplex processing is performed normally. The bit monitor 29 demodulates the RF signal from the RF combiner 37, and the controller 30 determines whether or not the RF signal is normal. The self-diagnosis of the demultiplexer 28 and the bit monitor 29 (judgment as to whether or not it is normal) is performed, for example, by using an error correction code (such as a BCH code) added when multiplexed by the multiplexer 25 (25A, 25B, 25C). The calculation may be performed, and the determination may be made based on the presence or absence of an error, an error amount, and the like.

For the diagnosis of the A / D converter 21, the output signal from the A / D converter 21 is
The operation may be confirmed by monitoring at 0. Signal compression circuit 2
For 3, switch to the compulsory self-diagnosis mode (test mode) to output, for example, a 1 kHz sine (sine) wave, return this to the bit monitor 29, and monitor the signal with the controller 30. The diagnosis may be made by

The self-diagnosis is performed by forcibly switching to the self-diagnosis mode when the power is turned on or at the time of maintenance as required, and the self-diagnosis using the error correction code added at the time of multiplexing is performed. The operation is performed periodically at a predetermined period even during the normal operation. When a failure occurs, for example, the failure status and the time of occurrence are written in the non-volatile memory 30M, and the history is stored.

The output level of the RF output signal from the RF modulator 122 can be controlled by the controller 30 by controlling the gain of the RF amplifier 36. The controller 30 is provided with, for example, a so-called EIA-485 interface, through which the current failure status and failure history can be monitored, and the output level of the RF signal can be controlled.

Next, the configuration of the seat unit 140 for receiving the transmitted audio signal, particularly the configuration after the RF distributor 144 in FIG. 3 will be described in detail with reference to FIG.

In FIG. 5, the RF signal split by the RF splitter 144 shown in FIG. 1 is input to the input terminal 41, and is split by the RF splitter 144 into three RF signals.
The signals are sent to the input terminals 42A, 42B, 42C of the three RF receivers 145A, 145B, 145C, respectively. Since the internal configurations of these RF receiving units 145A, 145B, and 145C are all the same, the internal configuration is illustrated only for one RF receiving unit, for example, 145A, and the other RF receiving units are illustrated.
Illustration of the internal configuration of the receiving units 145B and 145C is omitted. In addition, the suffix A is omitted from the designation code of each unit in the RF receiving unit 145A.

The RF signal input from the input terminal 42A of the RF receiving section 145A is sent to the tuner 52 of the RF receiving and demodulating circuit section 51. The tuner 52 includes the audio signal transmission units 121A, 121B, and 12 shown in FIG.
An RF signal of one band is selected (tuned and tuned) from the RF signals frequency-multiplexed at different RF carrier frequencies for each of 1C and 121D, and converted into an IF (intermediate frequency) signal. send. The demodulation circuit 53
The signal is demodulated to output serial data at the data rate of 6.144 MHz, which is sent to the demultiplexer 54.

As described above, the serial data supplied to the demultiplexer 54 is time-division multiplexed into three (3 frames) of the audio format (format conforming to the television signal of the satellite broadcast or the satellite communication). It was done. The demultiplexer 54 extracts the data corresponding to one frame by demultiplexing the time-division multiplex, performs error correction decoding processing (for example, BCH decoding processing) for each frame, and then sets a desired audio channel or data channel. Select and take out. The error correction processing is for detecting and correcting an inversion error of data which may be generated due to the influence of noise or the like at the time of transmitting / receiving the RF signal. The selection of the desired audio channel or data channel is performed according to a command from a microprocessor (microcomputer) 43.

Here, when the above time-division multiplexing is solved to extract only one frame signal, the audio channel or data channel is selected from 12 audio channels or 8 audio channels and 8 data channels in one frame. In the demultiplexer 54 of this embodiment, two channels of audio can be arbitrarily selected from 32 channels of audio and 8 channels of data corresponding to 3 frames, or the data can be arbitrarily selected. It is configured so that one channel can be selected. This can be achieved by extracting a signal corresponding to any two frames when, for example, solving the time-division multiplexing.

The data and clock selected by the demultiplexer 54 are taken out through buffer amplifiers. The audio signals of the two channels (for example, stereo left and right channels) selected by the demultiplexer 54 are sent to a D / A converter 56 via a decompression circuit 55 or not.

The decompression processing in the decompression circuit 55 is controlled so that switching between a first decompression mode corresponding to the first compression mode and a second decompression mode corresponding to the second compression mode can be controlled. The mode switching and the volume (volume) setting are performed in accordance with an instruction from the microprocessor 43.

The D / A converter 56 receives, for example, 16-bit audio data expanded by the expansion circuit 55, outputs a 2-channel analog audio signal, and sends it to the headphone amplifier 57. Headphone amplifier 5
Reference numeral 7 denotes an amplifier for driving headphones 147 connected via the control unit 146 on the passenger side in FIG. A, and outputs signals from left and right output terminals 47A _L and 47A.
Retrieved via _R.

Next, the self-diagnosis function of such a circuit will be described.

First, the microprocessor 43 sets the CP
The function diagnosis of each unit such as U, RAM, ROM and so-called watchdog timer is performed. If any of the tuning by the tuner 52, the demodulation by the demodulation circuit 53, and the error processing by the demultiplexer 54 have any trouble or failure,
The result (error detection output, etc.) is stored in the microprocessor 4
3 is read and stored in a memory or the like. This is an error correction code (such as the above-mentioned BCH code) previously added to the transmission signal.
Can be checked by reading the error flag, writing the result in a memory or the like, and controlling, for example, a so-called mute state, or a control not to cancel the mute if the current state is a mute state. You may.

At the time of audio self-diagnosis, the expansion circuit 55 outputs sine wave test data of, for example, 1 kHz, and the output from the headphone amplifier 57 is sent to the microprocessor 43 via the D / A converter 56. The microprocessor 43 has a built-in A / D
Detect with the converter and confirm that each circuit operates normally. Further, a 0 level signal is output from the expansion circuit 55,
The microprocessor 43 monitors the output from the so-called zero input detection terminal of the D / A converter 56,
It is checked whether the / A converter 56 is operating normally.

The self-diagnosis is performed by forcibly switching to the self-diagnosis mode at the time of power-on or maintenance, for example. The diagnosis by the error detection by the demultiplexer 54 is periodically performed at a predetermined period even during the normal operation. It may be performed on demand or as needed. Further, the result can be taken out through the input / output terminal 45. In addition,
The microprocessor 43 can exchange data with an external circuit via the input / output terminal 45.

Next, a so-called ATRAC system which can be used as the first compression mode will be described.

FIG. 6 shows a main configuration of the compression encoder (encoder) side. An audio signal input to an input terminal 61 is separated into a high-frequency component and a middle-low frequency component by an analysis filter 62. The low-frequency component is further separated by the analysis filter 63 into a middle-frequency component and a low-frequency component. The high frequency component from the analysis filter 62 is sent to an MDCT (improved discrete cosine transform) circuit 65H via a delay circuit 64. The mid-range component and the low-range component from the analysis filter 63 are respectively MDCT
It is sent to circuits 65M and 65L. Analysis filter 63,
A signal of each component from the block size determination circuit 6
6 to obtain a block size determination signal for the above-mentioned improved discrete cosine transform.
H, 65M and 65L. The data signals on the frequency axis output from the respective MDCT circuits 65H, 65M, 65L are taken out via output terminals 67H, 67M, 67L, respectively. Data compression is performed on these output signals on the frequency axis by thinning out data based on human auditory characteristics (particularly the minimum audibility characteristics and the auditory masking effect).

FIG. 7 shows the main structure of the ATRAC decompression decoder (decoder). The data signals on the frequency axis corresponding to the output signals from the output terminals 67H, 67M, 67L in FIG. 6 are supplied to the input terminals 71H, 71M, 71L in FIG.
The signals are sent to IMDCT (inverse improved discrete cosine transform) circuits 72H, 72M, and 72L, which perform inverse transform of CT.
Output signals from the IMDCT circuit 72H are sent to a delay circuit 73, and output signals from the IMDCT circuits 72M and 72L are sent to a synthesis filter 74. Further, the output signal from the delay circuit 73 and the output signal from the synthesizing filter 74 are sent to the synthesizing filter 75 and subjected to synthesizing processing.

The present invention is not limited to only the above-described embodiment. For example, the first compression mode may be a PASC adopted in a so-called dcc (digital compact cassette) other than the ATRAC system. (P
recision Adaptive Sub-band Coding)
Various high-efficiency compression encoding methods that can maintain high sound quality even when processing time is required can be used. Further, the system applied is not limited to the audio listening system in the aircraft, but can be applied to the audio listening system in other vehicles.

[0065]

As is apparent from the above description, according to the present invention, the first compression mode which requires a predetermined time for the signal compression processing, and the signal compression processing time is shorter than the first compression mode. The short second compression mode is switched and selected according to the signal content, for example, an audio signal such as music or an audio signal such as an announcement. Even if it takes a little longer time to perform the high-efficiency compression encoding process in the first compression mode with less sound quality degradation, and if it is desired to reduce the time delay such as in announcements, the processing in the second compression mode having a short processing time is performed. Thus, the optimum compression mode according to the signal content can be selected, and the signal can be transmitted in the compression mode suitable for the signal to be transmitted.

Further, according to the present invention, the first decompression mode which requires a predetermined time for signal decompression processing and the second decompression mode in which the signal decompression processing time is shorter than the first decompression mode, Since the switching is selected according to the content, for example, an audio signal such as music or an audio signal such as an announcement,
For applications requiring high sound quality, such as listening to music, perform a decompression process in the first decompression mode with little deterioration in sound quality even if the processing time is slightly longer, and if you want to reduce the time delay as in announcements, etc. By performing the processing in the second extension mode with a short time, an optimal extension mode according to the signal content can be selected, and the signal can be received in the extension mode suitable for the signal to be received.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a schematic configuration of a signal transmission system to which an embodiment according to the present invention is applied;

FIG. 2 is a diagram for explaining time and delay time required for processing in the signal transmission system of FIG. 1;

FIG. 3 is a block diagram showing a specific example of an audio signal transmission system to which the signal compression device according to the present invention is applied.

FIG. 4 is a block diagram showing a specific example of an audio signal transmission unit 121 in the audio signal transmission system shown in FIG.

FIG. 5 is a block diagram showing a specific example of a seat unit side for receiving an audio signal in the audio signal transmission system shown in FIG. 7;

FIG. 6 is a block circuit diagram showing a main configuration of a specific example of a high-efficiency compression encoder that can be employed as a first compression mode.

FIG. 7 is a block circuit diagram showing a main configuration of a specific example of a decompression side corresponding to a first compression mode.

[Explanation of symbols]

1 input terminal, 2 A / D converter, 3a, 3b
Compression circuit, 4,10 changeover switch, 5 transmission circuit, 8 reception circuit, 9a, 9b expansion circuit,
11 D / A converter

──────────────────────────────────────────────────続 き Continued on the front page (58) Field surveyed (Int. Cl. ⁷ , DB name) H03M 7/30 G10L 19/00

Claims (10)

(57) [Claims] 1. A first method which requires a predetermined time for signal compression processing.
And compression mode, signal compression processing time is a signal compressor having a switchable signal compression section between the shorter than the first compression mode in the second compression mode, as the signal content to be compressed, music Etc audio
Audio signals, such as announcements and other audio signals,
Signal is compressed in the first compression mode, and the audio signal is compressed in the first compression mode.
2. A signal compression apparatus according to claim 1, wherein the first compression mode and the second compression mode are switched and selected so as to perform compression in a second compression mode. 2. The first compression mode is a mode for performing high-efficiency coding that adaptively controls bit allocation for each band divided in the frequency axis direction using human auditory characteristics. The signal compression device according to claim 1, wherein the second compression mode is an adaptive difference PCM mode. 3. A signal expansion switchable between a first expansion mode that requires a predetermined time for signal expansion processing of a compressed signal and a second expansion mode in which the signal expansion processing time is shorter than the first expansion mode. A signal decompression device having a section, wherein the content of a signal to be decompressed is audio such as music.
Audio signals, such as announcements and other audio signals,
Signal is expanded in the first expansion mode, and the audio signal is expanded in the first expansion mode.
So as to extend in two stretching modes, signal expansion device of the compressed signal, characterized by switching selecting the said first decompression mode and a second decompression mode. 4. The first decompression mode is a mode for performing high-efficiency encoding that adaptively controls bit allocation for each band divided in the frequency axis direction using human auditory characteristics. 4. The signal decompression device according to claim 3 , wherein the second decompression mode is an adaptive difference PCM mode. 5. A signal transmission device comprising a signal compression device, wherein the signal compression device compresses a signal and transmits the signal via a transmission path, wherein the signal compression device requires a predetermined time for signal compression processing. a compression mode, as the signal content signal compression processing time has a switchable signal compression section between the shorter than the first compression mode in the second compression mode, to be compressed, the audio such as music
Audio signals, such as announcements and other audio signals,
Signal is compressed in the first compression mode, and the audio signal is compressed in the first compression mode.
A signal transmission device , wherein the first compression mode and the second compression mode are switched and selected so as to perform compression in a second compression mode. 6. The first compression mode is a mode for performing high-efficiency encoding that adaptively controls bit allocation for each band divided in the frequency axis direction using human auditory characteristics. The signal transmission device according to claim 5 , wherein the second compression mode is an adaptive difference PCM mode. 7. A signal receiving apparatus comprising a signal decompression device that receives a compressed signal via a transmission path and decompresses the compressed signal, wherein the signal decompression device requires a predetermined time for signal decompression processing of the compressed signal. A signal decompression device having a signal decompression unit that can be switched between a first decompression mode and a second decompression mode in which a signal decompression processing time is shorter than the first decompression mode, wherein the signal content to be decompressed is Audio, music, etc.
Audio signals, such as announcements and other audio signals,
Signal is expanded in the first expansion mode, and the audio signal is expanded in the first expansion mode.
A signal receiving apparatus , wherein the first and second expansion modes are switched and selected so as to perform expansion in a second expansion mode. 8. The first decompression mode is a mode for performing high-efficiency encoding that adaptively controls bit allocation for each band divided in the frequency axis direction using human auditory characteristics. The signal receiving apparatus according to claim 7 , wherein the second expansion mode is an adaptive difference PCM mode. 9. A signal transmission device comprising a signal compression device, which compresses a signal with the signal compression device and transmits the signal via a transmission path;
A signal transmission / reception device that receives a compressed signal via a transmission path and expands the compressed signal, wherein the signal compression device requires a predetermined time for signal compression processing.
The first compression mode and the signal compression processing time
Possible changeover between shorter than the compression mode the second compression mode
The signal decompression device includes a signal decompression unit , and the signal decompression device includes a first decompression mode that requires a predetermined time for signal decompression processing of the compressed signal, and a second decompression time that is shorter than the first decompression mode. It has a signal decompression unit that can be switched between modes, and the signal content to be compressed or decompressed includes music, etc.
Using audio signals and audio signals such as announcements,
Audio signals are compressed in the first compression or decompression mode
Or the audio signal is expanded in the second compression or expansion mode.
In to compress or stretch, the a first compression or decompression mode, the signal transmitting and receiving apparatus, characterized by switching selecting the said second compression or decompression mode. 10. The first compression or decompression mode is a mode for performing high-efficiency encoding that adaptively controls bit allocation for each band divided in the frequency axis direction using human auditory characteristics. 10. The signal transmitting / receiving apparatus according to claim 9 , wherein the second compression or decompression mode is an adaptive difference PCM mode.